System and method for transmitting and receiving wideband speech signals

ABSTRACT

The system for transmitting and receiving a wideband speech signal includes an A/D converter for receiving an analog speech signal to convert it into a digital speech signal, a transmitter analysis filter for receiving the digital speech signal and dividing it into a baseband signal and an enhancement residual band signal, a standard baseband encoder for accepting the baseband signal and coding it using an ITU-T encoder, an additional baseband encoder for reducing standard coding distortion in the baseband signal, an enhancement residual band encoder for coding a signal obtained by removing the coded baseband signal from the original digital speech signal, and an IP network interface for multiplexing the coded standard and additional baseband signals and enhancement residual band signal.

CROSS REFERENCE TO RELATED APPLICATION

This application is based on Korea Patent Application No. 2002-81663filed on Dec. 20, 2002 in the Korean Intellectual Property Office, thecontent of which is incorporated herein by reference.

BACKGROUND OF THE INVENTION

(a) Field of the Invention

The present invention generally relates to a system and method fortransmitting and receiving wideband speech signals.

(b) Description of the Related Art

A wideband speech signal has a frequency of 0˜8 kHz, which is twice thatof a currently used telephone-line band, at 0˜4 kHz. The wideband speechsignal is suitable for a next-generation speech communication systembecause it is less artificial than a signal of the telephone-line band,and has high intelligibility.

The G.711 is one of the methods of digitalizing and compressing a speechsignal of the telephone-line band, which has been standardized by ITU-T(International Telecommunication Union-Telecommunication Standardizationsector) for the first time. G.711 uses nonlinear pulse code modulation(u/A-PCM), and it is known that its performance is similar to theoriginal speech signal at 64 kbit/s. Other speech compression methodsstandardized by ITU-T include G.726 (adaptive differential pulse codemodulation, ADPCM, 32 kbit/s), G.728 (low delay code excited linearprediction, LD-CELP, 16 kbit/s), G.729 (conjugate structure algebraiccode excited linear prediction, CS-ACELP, 8 kbit/s), and G.723.1(ACELP/MP-MLQ, 5.3/6.3 kbit/s).

Among ITU-T standard speech coders, G.723.1 and G.729 were adopted as aVoIP (Voice over Internet Protocol) standard coder on the Internet.G.723.1, developed for the purpose of compressing multimedia signals ata low rate, is an algorithm capable of compressing and decompressinginput speech at two bit rates of 5.3 kbit/s and 6.3 kbit/s. G.723.1 usesthe analysis-by-synthesis method, which has the widest application fieldamong speech coding methods, and provides toll quality as high as thatof a wired network. ITU-T G.729 is an algorithm capable of compressingand decompressing input speech at the rate of 8 kbit/s using analgebraic codebook, and it also provides toll quality as high as that ofthe wired network. Furthermore, ITU-T G.729A (G.729 Annex A) has thesame transmission parameter as that of G.729 so it has compatibilitywith G.729 and it has advantages in complexity. Accordingly, G.729A iswidely used in actual systems.

The above-described speech coding methods or systems with lowtransmission rates that provide toll quality as high as that of wirednetworks have created new services in mobile communications and Internetphone services. In particular, VoIP on the Internet is being spread veryrapidly due to inexpensive telephone charges. However, theaforementioned conventional coding methods or systems have had poorservice quality because of low quality and long delay time caused bypacket loss on the Internet. With the development of communicationnetworks and protocols, however, most of the problems in theconventional coding methods or systems with respect to delay timegenerated on networks have been solved. Recently, attempts to extend aspeech signal to a wideband signal having a frequency (0˜8 kHz) twicethat of telephone-line band (0˜4 kHz) to improve toll quality have beenmade. However, the 16 kHz wideband speech codec, which has been alreadystandardized, has no backward compatibility with the telephone-line bandcodec currently being used for VoIP services so a new communicationsystem must be designed in order to use signals of the two differentsystems. Furthermore, since the wideband signal occupies a widebandwidth, a network capable of processing a large amount of data isneeded. Accordingly, there are many problems in providing new serviceswith the current systems.

SUMMARY OF THE INVENTION

An object of the present invention is to provide a system and method fortransmitting and receiving a wideband speed signal, which is compatiblewith the conventional systems and provides a sampled 16 kHz widebandspeech signal with high quality.

To accomplish the object of the present invention, there is provided asystem for transmitting a wideband speech signal, comprising an A/Dconverter for receiving an analog speech signal to convert it into adigital speech signal; an acoustic echo canceller for receiving thedigital speech signal and canceling echo of the digital speech signal;an encoder for receiving the speech signal from which echo has beenremoved from the acoustic echo canceller to separate a baseband speechsignal from the received signal, standard-coding the baseband speechsignal to generate a standard baseband signal, coding a differencebetween the baseband speech signal and the standard baseband signal togenerate an additional baseband signal, and coding a difference betweena signal obtained by synthesizing the standard baseband signal and theadditional baseband signal and the baseband speech signal to generate anenhancement residual band signal; and an IP network interface forreceiving the standard baseband signal, the additional baseband signal,and the enhancement residual band signal generated by the encoder,multiplexing them, and transmitting the multiplexed signals to anexternal network connected to the transmitting system.

To accomplish the object of the present invention, there is alsoprovided a system for receiving a wideband speech signal, comprising anIP network interface for demultiplexing a speech signal transmittedthrough a network to extract a coded standard baseband signal, a codedadditional baseband signal, and a coded enhancement residual bandsignal; a decoder for standard-decoding the standard baseband signal,decoding the additional baseband signal, and synthesizing the decodedsignals to generate a baseband synthesis signal, and decoding theenhancement residual band signal and synthesizing the decodedenhancement residual band signal and the baseband synthesis signal togenerate a final decoded speech signal; an acoustic echo canceller forcanceling echo of the decoded speech signal; and a D/A converter forreceiving the speech signal from which echo has been removed toconverting it into an analog signal.

BRIEF DESCRIPTION OF THE DRAWINGS

Further objects and advantages of the invention can be more fullyunderstood from the following detailed description taken in conjunctionwith the accompanying drawings, in which:

FIG. 1 shows a communication network to which a transmitter and areceiver according to the present invention are applied;

FIG. 2 shows the configuration of the transmitter according to thepresent invention;

FIG. 3 shows the encoder of the transmitter of FIG. 2 in detail;

FIG. 4 is a flow chart showing a transmission method according to thepresent invention;

FIG. 5 shows the configuration of the receiver according to the presentinvention;

FIG. 6 shows the decoder of the receiver of FIG. 2 in detail;

FIG. 7 is a flow chart showing a receiving method according to thepresent invention;

FIG. 8 shows a transceiver according to the present invention; and

FIG. 9 shows the structure of a general VoIP service protocol.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT

The present invention will now be described in connection with preferredembodiments with reference to the accompanying drawings.

FIG. 1 shows a communication network to which a transmitter and areceiver according to the present invention are applied. In theconfiguration shown in FIG. 1, a network 104 that secures quality ofservice (QoS) and a network 103 that does not guarantee quality ofservice are simultaneously provided. In this configuration, IP terminals101 and 106 represent terminals that are capable of generating andtransmitting IP packets. Each IP terminal includes a microphone and aspeaker that can receive and output speech signals, and a terminalthrough which a user can make a phone call. The IP terminal can berealized in various ways, using a telephone or a personal computer forinstance. In the case that packets generated by one IP terminal 101 aretransmitted to another IP terminal 106, the packets are divided intostandard baseband packets 105, additional baseband packets, andenhancement residual band packets 102. In FIG. 1, the number attached toeach of packets 102 and 105 represents a time stamp.

The standard baseband packets 105 are transmitted through the network104 that guarantees quality of service because they hold importantinformation required for speech communication. The additional basebandpackets can be sent when a quality of service channel has sufficientcapacity. In the case that only the baseband is used, intelligibilityand naturalness of speech are considerably deteriorated although meaningis transmitted to a counterpart in speech communication. Accordingly, auser who wants speech quality corresponding to face-to-facecommunication can use high speech-quality communication when decoding iscarried out even using the enhancement residual band packets transmittedthrough the network 103 that does not secure quality of service. Sincethe packets transmitted through the network 103 not securing quality ofservice have irregular delays, packet loss, and jitter, the IP terminal106 corresponding to a receiving part uses a device for improvingquality of service, such as a jitter buffer, FEC (frame error control)device, or the like, for synchronization of packets so as to maximizespeech quality. In the case that a variation in the channel capacity ofa network can be detected in real time, frame size can be varied withtime.

A wideband speech signal transmitter according to a preferred embodimentof the present invention is explained below with reference to FIG. 2 andFIG. 3. FIG. 2 shows the configuration of the transmitter according tothe present invention, and FIG. 3 shows the encoder of the transmitterof FIG. 2 in detail.

As shown in FIG. 2, the transmitter of the invention includes an A/Dconverter 201, an acoustic echo canceller 202, an encoder 203, and an IPnetwork interface 204. The A/D converter 201 receives an analog speechsignal of a user through a microphone to convert it into a digitalspeech signal. The acoustic echo canceller 202 cancels echo in thespeech signal received from the A/D converter 201. It is possible to usean acoustic echo canceller standardized by ITU-T G.167 as the acousticecho canceller of the invention. In the embodiment of the presentinvention, the signal applied to and outputted from the acoustic echocanceller 202 has a frequency of 0˜8 kHz, and it is a 16-bit linear PCMsignal having a sampling rate of 16 kHz.

The encoder 203 receives the signal from which the echo has beenremoved, from the acoustic echo canceller 202, and codes a basebandsignal and an enhancement residual band signal. The enhancement residualband signal is obtained from a difference between the original signaland the baseband signal. A detailed configuration of the encoder 203 isshown in FIG. 3, which will be explained later in more detail.

The IP network interface 204 multiplexes the input signal consisting ofthe standard baseband signal, the additional baseband signal, and theenhancement residual band signal in various manners according to thestructure of an IP network to efficiently transmit packets. The IPnetwork interface 204 can have various options according to thestructure of an external network connected thereto. The options are asfollows. First, if all of networks provided for speech communicationguarantee quality of service, the IP network interface simultaneouslypacketizes the standard baseband signal, the additional baseband signal,and the enhancement residual band signal, or packetizes only thestandard baseband signal and the additional baseband signal to transmitthem. Secondly, in the case that a network that secures quality ofservice and a network that does not guarantee quality of service aresimultaneously connected to the network interface, the network interfacetransmits the standard baseband signal and the additional basebandsignal, which are relatively important information, through the networkthat secures quality of service and delivers the enhancement residualband signal through the network that does not guarantee quality ofservice. This case provides speech quality much better than thatobtained in the case of using only the baseband signals even when packeterror or jitter is generated on the network that does not secure qualityof service. Finally, when all networks connected to the networkinterface do not guarantee quality of service, the network interface 204simultaneously packetizes the baseband signals and enhancement residualband signal or packetizes only the baseband signals to transmit them asin the first case.

The IP network interface 204 generates packets in the form as shown inFIG. 9. While the VoIP currently being provided is constructed of acombination of various protocols, the present invention can be appliedto any combination of protocols described in FIG. 9.

FIG. 3 illustrates the encoder 203 shown in FIG. 2 in more detail.Referring to FIG. 3, the encoder 203 includes a transmitter analysisfilter 301, a standard baseband encoder 302, a standard baseband decoder303, an additional baseband encoder 304, an additional baseband decoder305, and an enhancement residual band encoder 306. The transmitteranalysis filter 301 receives a digital speech signal that is a 16-bitlinear PCM signal having a bandwidth of 0˜8 kHz sampled at 16 kHz, tooutput a baseband speech signal having a bandwidth of 0˜4 kHz at thesampling rate of 8 kHz. The standard baseband encoder 302 accepts thebaseband speech signal and encodes it according to G.723.1 or G.729A,which are standard encoding methods, to output a coded standard basebandsignal. The additional baseband encoder 304 receives a difference signalbetween a filtered baseband speech signal and the signal decoded by thestandard baseband decoder 303 and codes it through waveform coding ortransform coding, to generate an additional baseband signal. Theenhancement residual band encoder 306 up-samples a signal, acquired bysynthesizing the signals outputted from the standard baseband decoder303 and additional baseband decoder 305, at 16 kHz, obtains a differencebetween the up-sampled signal and the speech signal applied to thetransmitter analysis filter 301, and applies waveform coding ortransform coding to the obtained difference signal, to generate anenhancement residual band signal. Here, a method of using human acousticcharacteristics can be used. Next, a method of transmitting a widebandspeech signal according to an embodiment of the present invention isexplained with reference to FIG. 4. FIG. 4 is a flow chart showing thetransmission method according to the present invention.

When the transmission operation starts, an analog speech signal isreceived and converted into a digital signal at step 401. Then, echo inthe digital speech signal is cancelled at step S402. Subsequently, thespeech signal from which echo has been removed passes through a low passfilter to obtain only a baseband signal, and this baseband signal iscoded by the standard baseband encoder, at step 403. In preparation fordeterioration in the performance of the transmitter due to the standardcoding, the additional baseband encoder codes the baseband speech signalat step 404. At step 405, the sampling frequency of a signal obtained bydecoding a standard baseband signal and an additional baseband signal isdoubled, and then an enhancement residual band signal is obtained from adifference between the signal having the doubled sampling frequency andthe original 16 kHz speech signal from which echo has been cancelled andcoded. At step 406, the coded speech signals are multiplexed in variousmanners according to the kind of network connected to the transmitter.The multiplexed signals are packetized and transmitted through thenetwork at step S407.

As described above, the digital speech signal converted from the analogspeech signal is a 16-bit linear PCM signal and has a sampling rate of16 kHz. In addition, each of the standard baseband signal and theadditional baseband signal can have a sampling rate of 8 kHz and abandwidth of 0˜4 kHz or 4˜8 kHz. Furthermore, the standard basebandsignal conforms to the G.723.1 or G.729A coding technique, and waveformcoding or transform coding that is applied to the standard basebandsignal can be used for the additional baseband signal. The enhancementresidual band signal can also be coded through waveform coding ortransform coding.

The wideband speech signal receiver according to an embodiment of thepresent invention is described with reference to FIG. 5 and FIG. 6. FIG.5 shows the wideband speech signal receiver of the present invention andFIG. 6 illustrates the decoder shown in FIG. 5 in more detail.

Referring to FIG. 5, the wideband speech signal receiver of theinvention includes an IP network interface 501, a decoder 502, anacoustic echo canceller 503, and a D/A converter 504. The IP networkinterface 501 receives a multiplexed standard baseband signal,additional baseband signal, and if required, the enhancement residualband signal, and demultiplexes them, to extract coded baseband speechsignals or an enhancement residual band speech signal. The decoder 502accepts the baseband speech signals and enhancement residual band speechsignal extracted by the IP network interface 501 to decode them, andsynthesizes the baseband speech signals and enhancement residual bandspeech signal into one speech signal. Here, the decoding method isidentical to the above-described coding method. The detailedconfiguration of the decoder will be explained later with reference toFIG. 6.

The acoustic echo canceller 503 receives the synthesized speech signalfrom the decoder 502 to cancel echo. As the acoustic echo canceller ofthe present invention, an acoustic echo canceller standardized by ITU-TG.167 can be used. The D/A converter 504 accepts the speech signal fromwhich the echo has been removed to convert it into an analog signal.This analog signal can be provided to a user through a speech signaloutput unit.

The decoder 502 is explained in more detail with reference to FIG. 6.Referring to FIG. 6, the decoder 502 includes an enhancement residualband decoder 601, a standard baseband decoder 602, an additionalbaseband decoder 603, and a receiver synthesis filter 605. Theenhancement residual decoder 601 receives the coded enhancement residualband speech signal transmitted from the transmitter, and decodes theenhancement residual band speech signal through waveform decoding ortransform decoding. The standard baseband decoder 602 accepts the codedbaseband speech signal and decodes it according to G.723.1 or G.729A, tooutput a baseband speech signal. The additional baseband decoder 603receives the additional baseband signal to decode it. The signal decodedby the additional baseband decoder 603 and the signal from the standardbaseband decoder 602 are synthesized to produce a final basebandsynthesis signal. The receiver synthesis filter 604 synthesizes thesignal outputted from the enhancement residual band decoder 601 and thebaseband synthesis signal, to obtain a finally decoded speech signal.

A method of receiving a wideband speech signal according to anembodiment of the present invention with reference to FIG. 7 will now beexplained. FIG. 7 is a flow chart showing the receiving method.

When the receiving operation starts, a speech signal multiplexedaccording to the kind of network is received at step 701. The receivedsignal is demultiplexed to be divided into an enhancement residual bandsignal and a baseband speech signal at step 702. The baseband speechsignal consists of a standard baseband signal and an additional basebandsignal. The enhancement residual band signal and the baseband speechsignals are respectively decoded at step 703. The decoded enhancementresidual band signal and baseband speech signals are synthesized into asingle speech signal at step 704. Subsequently, echo in the synthesizedspeech signal is cancelled at step 705. The speech signal obtained atstep 705 is converted into an analog signal at step 706. Here, thespeech signal decoding method is identical to the decoding methodexplained in FIG. 5 and FIG. 6.

FIG. 8 shows a preferred configuration of a transceiver system in whichthe transmitter and receiver for transmitting and receiving widebandspeech signals according to the present invention are integrated into asingle terminal.

Referring to FIG. 8, an A/D converter 801 receives an analog speechsignal from a user through a microphone and converts it into a digitalspeech signal, then an acoustic echo canceller 802 cancels echo in thedigital speech signal received from the A/D converter 801. In addition,the acoustic echo canceller 802 accepts a received digital speech signalto cancel echo included therein. An enhancement residual band encoder804 receives an enhancement residual band signal from a transmitterfilter 803 to code it through waveform coding or transform coding andoutput the coded signal to an IP network interface 807.

A baseband encoder is divided into a standard baseband encoder 805 andan additional baseband encoder 806. The standard baseband encoder 805accepts the baseband signal from the transmitter filter 803 to code itaccording to G.723.1 or G.729A, and outputs the coded signal to thenetwork interface 807. The additional baseband encoder 806 codes thesignal coded by the standard baseband encoder 805 through waveformcoding or transform coding in order to reduce distortion in the signal,and then outputs the coded signal to the network interface 807.

The IP network interface 807 respectively receives the coded enhancementresidual band signal and the coded standard and additional basebandsignals from the enhancement residual encoder 804, the standard basebandencoder 805, and the additional baseband encoder 806 to multiplex them.Here, the enhancement residual band signal and the standard andadditional baseband signals are multiplexed in various combinationmanners according to the kind of network through which the signals areto be transmitted. The multiplexed signal is delivered to the networkconnected to the network interface 807. In the case that all thenetworks provided for speech signal communication secure quality ofservice, the baseband signals and enhancement residual band signal aresimultaneously packetized or only the baseband signals are packetized tobe transmitted. When a network that guarantees quality of service and anetwork that does not secure quality of service are simultaneouslyprovided, the baseband speech signal that is relatively importantinformation is transmitted through the network that guarantees qualityof service and the enhancement residual band signal is delivered throughthe network that does not secure quality of service. Here, even in thecase that packet error or jitter is generated on the network that doesnot secure quality of service, speech quality is much better than thatobtained in the case of using only the baseband speech signal. Finally,if all the networks do not guarantee quality of service, the basebandspeech signals and enhancement residual band signal are simultaneouslypacketized or only the baseband speech signals are packetized to betransmitted, as in the first case.

The IP network interface 807 receives a multiplexed signal from anetwork obtained by multiplexing the baseband speech signals orenhancement residual band signal according to the kind of network thesignal is received from, and demultiplexes it to extract a codedenhancement residual band signal and a coded baseband speech signal. Anenhancement residual band decoder 808 receives the extracted codedenhancement residual band speech signal from the IP network interface807 to decode it according to waveform decoding or transform decoding.

A standard baseband decoder 809 and an additional baseband decoder 810accept the extracted baseband speech signal to standard-decode itaccording to G.723.1 or G.729A and decode an additional baseband signal.A receiver synthesis filter 811 receives the speech signals decoded bythe enhancement residual band decoder 808 and the baseband decoders 809and 810 to synthesize them into a single signal and output this signalto a D/A converter 812. The D/A converter 812 receives the speech signalwhose echo has been cancelled from the acoustic echo canceller 802 toconvert it into an analog signal. This analog signal can be provided toa user through a speech output unit.

The digital speech signal converted from the analog speech signalinputted by a user is a 16-bit linear PCM signal and has a sampling rateof 16 kHz. The baseband speech signal has a sampling rate of 8 kHz and abandwidth of 0˜4 kHz. The enhancement residual band signal has asampling rate of 16 kHz and a bandwidth of 0˜8 kHz, or it has a samplingrate of 8 kHz and a bandwidth of 4˜8 kHz.

The transmitting and receiving methods according to the embodiments ofthe present invention can be constructed of a program executable on acomputer and they can be realized on a general digital computeroperating the program using a recording medium readable by the computer.The recording medium includes a magnetic storage medium (for example,ROM, floppy disk, hard disk, etc.), an optically read medium (forinstance, CD-ROM, DVD, etc.) and a carrier wave (for example,transmission through the Internet).

Although specific embodiments including the preferred embodiment havebeen illustrated and described, it will be obvious to those skilled inthe art that various modifications may be made without departing fromthe spirit and scope of the present invention, which is intended to belimited solely by the appended claims.

As described above, the system of the present invention is compatiblewith the current coding method when only signals using only the standardbaseband encoder are utilized according to a user's choice. Thus, thereis no need to use a new system. Furthermore, high quality speechcommunication can be carried out if the additional baseband signal andenhancement residual band signal are used in addition to the standardbaseband signal. Moreover, speech signals are multiplexed in variousmanners according to the kind of network and transmitted through thenetwork so that the speech signals can be efficiently delivered. Inaddition, in order to reproduce high sound quality, echo is cancelled soa loud speaker may be used. This produces a better communicationenvironment.

1. A system for transmitting a wideband speech signal, comprising: anA/D converter for receiving an analog speech signal to convert theanalog speech signal into a digital speech signal; an acoustic echocanceller for receiving the digital speech signal and canceling echo ofthe digital speech signal; an encoder to receive the digital speechsignal from which echo has been removed from the acoustic echo cancellerto separate a baseband speech signal from the received signal,standard-coding the baseband speech signal to generate a standardbaseband signal, coding a difference between the baseband speech signaland the standard baseband signal to generate an additional basebandsignal, and coding a difference between a signal obtained bysynthesizing the standard baseband signal and the additional basebandsignal and the baseband speech signal to generate an enhancementresidual band signal; and an IP network interface to receive thestandard baseband signal, the additional baseband signal, and theenhancement residual band signal generated by the encoder, multiplexingthe standard baseband signal, the additional baseband signal, and theenhancement residual band signal, and transmitting the multiplexedsignals to an external IP network.
 2. The system for transmitting awideband speech signal as claimed in claim 1, wherein the encodercomprises: a transmitter analysis filter to receive the digital speechsignal and to divide the digital speech signal into the baseband speechsignal and the enhancement residual band signal; a standard basebandencoder for standard-coding the baseband speech signal to generate thestandard baseband signal; an additional baseband encoder for coding thedifference between the baseband speech signal and the standard basebandsignal, to generate the additional baseband signal; and an enhancementresidual band encoder for coding the difference between a signalobtained by synthesizing the standard baseband signal and the additionalbaseband signal, and the baseband speech signal, to generate theenhancement residual band signal.
 3. The system for transmitting awideband speech signal as claimed in claim 2, wherein the standardbaseband encoder codes the baseband speech signal according to G.723.1or G.729A, and the additional baseband encoder codes the differencebetween the baseband speech signal and the standard baseband signalthrough waveform coding or transform coding, the human acousticcharacteristic being reflected on the waveform coding or transformcoding.
 4. The system for transmitting a wideband speech signal asclaimed in claim 1, wherein the IP network interface simultaneouslypacketizes the enhancement residual band signal, the standard basebandsignal, and the additional baseband signal and transmits the enhancementresidual band signal, the standard baseband signal, and the additionalbaseband signal, or packetizes the standard baseband signal and theadditional baseband signal and transmits the standard baseband signaland the additional baseband signal, in the case that the networkconnected to the transmitting system includes a network that securesquality of service or a network that does not guarantee quality ofservice, and when the IP network connected to the transmitting systemincludes both the network that secures quality of service and thenetwork that does not guarantee quality of service, the IP networkinterface packetizes the standard baseband signal to transmit thepacketized additional baseband signal and enhancement residual bandsignal, the packetized standard baseband signal through the network thatsecures quality of service, and packetizes the additional basebandsignal and enhancement residual band signal to deliver them through thenetwork that does not guarantee quality of service.
 5. The system fortransmitting a wideband speech signal as claimed in claim 1, wherein,when the network connected to the transmitting system includes both thenetwork that secures quality of service and the network that does notguarantee quality of service, the IP network interface packetizes thestandard baseband signal and additional baseband signal to transmit thepacketized standard baseband signal and additional baseband signalthrough the network that secures quality of service, and packetizes theenhancement residual band signal to deliver the packetized enhancementresidual band signal through the network that does not guarantee qualityof service.
 6. The system for transmitting a wideband speech signal asclaimed in claim 1, wherein the digital speech signal outputted from theA/D converter is sampled at the rate of 16 kHz and has a frequency bandof 0-8 kHz, the enhancement residual band signal has a sampling rate of16 kHz and a frequency of 0-8 kHz, and the baseband speech signal has asampling rate of 8 kHz and a frequency band of 0-4 kHz.
 7. The systemfor transmitting a wideband speech signal as claimed in claim 1, whereinthe digital speech signal outputted from the A/D converter is sampled atthe rate of 16 kHz and has a frequency band of 0-8 kHz, the enhancementresidual band signal has a sampling rate of 8 kHz and a frequency bandof 0-4 kHz through down-sampling, and the baseband speech signal has asampling rate of 8 kHz and a frequency band of 0-4 kHz.
 8. A system forreceiving a wideband speech signal, comprising: an IP network interfacefor demultiplexing a speech signal transmitted through a network toextract a coded standard baseband signal, a coded additional basebandsignal, and a coded enhancement residual band signal; a decoder forstandard-decoding the standard baseband signal, to decode the additionalbaseband signal, and to synthesize the decoded signals to generate abaseband synthesis signal, to decode the enhancement residual bandsignal and synthesize the decoded enhancement residual band signal andthe baseband synthesis signal to generate a final decoded speech signal;an acoustic echo canceller cancel echo of the decoded speech signal; anda D/A converter for receiving the speech signal from which echo has beenremoved to convert the echo-removed speech signal into an analog signal.9. The system for receiving a wideband speech signal as claimed in claim8, wherein the decoder comprises: an enhancement residual band decoderto receive the enhancement residual band signal extracted by the IPnetwork interface and decode the enhancement residual band signalthrough waveform decoding or transform decoding; a standard basebanddecoder to receive the standard baseband signal extracted by the IPnetwork interface and decoding it according to G.723.1 or G.729A; anadditional baseband decoder receive the additional baseband signalextracted by the IP network interface and decoding the extractedadditional baseband signal; and a receiver synthesis filter forsynthesizing the signal outputted from the standard baseband decoder andthe signal outputted from the additional baseband decoder to generate abaseband signal, and synthesizing the baseband signal and theenhancement residual band signal to generate a finally decoded speechsignal.
 10. The system for receiving a wideband speech signal as claimedin claim 8, wherein the enhancement residual band signal has a samplingrate of 16 kHz and a frequency band of 0-8 kHz or a sampling rate of 8kHz and a frequency band of 4-8 kHz, and the baseband speech signal hasa sampling rate of 8 kHz and a frequency band of 0-4 kHz.